Fiber to the user (“FTTU’), fiber to the curb (“FTTC”), fiber to the node (“FTTN”), and fiber to the premise (“FTTP”) platforms (referred to herein as “FTTx”), require plain old telephony service (“POTS”) emulation using VoIP signaling and bearer channels. Such a POTS emulation service requires emulation of existing fault isolation mechanisms in POTS and also must be extended to the topologies in the VoIP environment. Some similar tests are available in the circuit switch based POTS. However, these conventional solutions do not address the POTS emulation services over data networks.
As used herein, “MEGACO” refers to the H.248 gateway control protocol, “MGCP” refers to the media gateway control protocol, and “SIP” refers to the session initiation protocol.
VoIP implementations enable users to carry voice traffic (for example, telephone calls and faxes) over an IP network. A VoIP system consists of a number of components including a gateway/media gateway, a gatekeeper, a call agent, a media gateway controller, a signaling gateway, application gateways, session border controllers, a call manager, and other components.
For example, a media gateway converts media provided in one type of network to the format required for another type of network. A gateway could terminate bearer channels from a switched circuit network and media streams from a packet network. This gateway may be capable of processing audio, video and T.120 alone or in any combination, and is capable of full duplex media translations.
VoIP technology utilizes a digital signal processor (“DSP”) to segment the voice signal into frames and store them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a data network using signaling protocols such as H.323, MGCP, MEGACO or SIP.
As VoIP is a delay and jitter sensitive application, a well-engineered end-to-end network is necessary to use VoIP successfully.